Methods and apparatus for conducting conference calls using a conferencing system adapted to operate between a circuit-switched network and a packet-switched network

ABSTRACT

The present invention is generally related to conference calls and more specifically, to methods and apparatus for conducting conference calls using a conference system adapted to operated between a circuit-switched network and a packet-switched network. In one embodiment, a method for conducting conference calls using a conferencing system is disclosed. In another embodiment, a conferencing system is disclosed.

CROSS REFERENCE TO RELATED APPLICATIONS

This application claims the benefit of the filing date of U.S.provisional patent application Ser. No. 60/552,469, filed on 12 Mar.2004 to the fullest extent permitted under 35 U.S.C. §119(e), and thecontents of such provisional patent application are incorporated by thisreference as if set forth verbatim herein.

The present invention is generally related to conference calls and morespecifically, to methods and apparatus for conducting conference callsusing a conference system adapted to operate between a circuit-switchednetwork and a packet-switched network.

SUMMARY OF THE INVENTION

The present invention provides methods and apparatus for conductingconference calls using a conference system adapted to operate between acircuit-switched network and a packet-switched network.

In one embodiment, a method for conducting conference calls using aconferencing system is disclosed. The method for conducting conferencecalls using a conferencing system is adapted to operate between acircuit-switched network and a packet-switched network. The methodcomprises the following: receiving on a VRU a plurality of requests forconferencing services from a plurality of callers, wherein at least someof the requests arrive via the circuit-switched network. A givenconference call is assigned involving at least two of the callers to agiven mixer coupled to communicate with the VRU via the packet-switchednetwork. Respective voice streams originating from respective some ofthe plurality of callers are mixed. Once the voice streams are mixed, amixed conference stream is routed from mixer to the VRU via thepacket-switched network and the mixed conference stream is routed tocallers in the given conference call from the VRU via thecircuit-switched network.

In another embodiment, a conferencing system is disclosed. Theconferencing system comprises at least the following: at least one voiceresponse unit that is adapted to interact via a circuit-switched networkwith a plurality of callers. The voice response unit is further adaptedto support at least one conferencing application and at least onenon-conferencing application. There is at least one mixer incommunication with the voice response unit via a packet-switchednetwork. The at least one mixer is adapted to support at least oneconference call between at least two callers communicating with oneanother via the mixer and the VRU and at least one data store is adaptedto store data representing at least one state parameter relating to atleast one conference call supported by the mixer. The data store iscoupled to communicate with the VRU and the mixer.

BRIEF DESCRIPTIONS OF THE DRAWING FIGURES

FIG. 1 is a diagram of an illustrative message sequence suitable forestablishing a new conference.

FIG. 2 is a diagram of an illustrative message sequence suitable foradmitting at least one additional conference to an existing conference.

FIG. 3 is a diagram of an illustrative message sequence suitable forexecution when a conference participant terminates a connection to theconference.

FIG. 4 is a diagram of an illustrative message sequence suitable forexecution when a conference host terminates a connection to theconference.

FIG. 5 is a diagram of an illustrative message sequence suitable forenabling a conference host to mute all conference participants.

FIG. 6 is a block diagram of an illustrative message sequence suitablefor specifying the announcement played to conferees upon entry to orexit from a conference call conducted according to the instantinvention.

FIG. 7 is a diagram of an illustrative message sequence suitable forlocking or unlocking a given conference call conducted according to theinstant invention.

FIG. 8 is a block diagram of various component and data flows related toa representative overall conference call flow provided according to theinstant invention.

DETAILED DESCRIPTION OF ILLUSTRATIVE EMBODIMENTS Overview

FIG. 8 is a block diagram illustrating components and data flowsassociated with a. Some of the data flows shown in FIG. 8 can beaccomplished using messages passed between the various components via aproxy server 115. However, to promote clarity and conciseness, and toavoid overcrowding in FIG. 8, some of these data flows are shown asbypassing the proxy server 115, when some of these flows may indeed passthrough the proxy server 115. Thus, the data flows appearing in FIG. 8(and all other drawings herein) are chosen for expository convenienceonly, and do not limit the instant invention.

To facilitate understanding of an illustrative but non-limiting sequenceof processing shown in FIG. 8, sequence numbers appear in FIG. 8 insidecircles, and these sequence numbers correspond to the paragraph numbersherein.

Caller into New Conference

1. Turning to FIG. 8, a caller 700 dials a predefined access number torequest access to a conference call. The caller 700 can be either aconference host or a conference participant, as those terms are used anddefined herein. The request from the caller 700 can be routed to a VRU705, which obtains a passcode or other host-specific conference code 711from the caller 700. The term “passcode” or “conference code” 711 refersto any unique identifier associated with the conference call host toidentify conferences associated with a given conference call host to theexclusion of conferences associated with all other conference callhosts. Having obtained the passcode/conference code 711 or otherhost-specific conference code, the VRU 705 can forward this code 711 toother components for further processing as discussed below. The caller700 can be prompted to enter a conference code 711 (e.g., manually orverbally through a communications device), or more generally, the VRU705 can prompt the caller 700 to enter a passcode, a conference code, orother conference configuration information. Typically, the caller 700will respond to these prompts by keying in the requested data via DTMFinput or by speaking the requested codes. In the latter case, a speechrecognition engine (readily available from a variety of vendors) may bedeployed to process speech input from the caller 700. Generally, if thecaller 700 responds with a conference code 711, that code can passdirectly to the VRU 705, or can pass through any number of intermediatecomponents between the VRU 705 and the caller 700.

2. The VRU 705 can forward the conference code 711 to a provisioningdatabase 710 for validity checking. The provisioning database 710 can bepopulated when conference hosts enroll for conferencing services andthus receive conferencing codes 711. The conference hosts in turnprovide conferencing codes or participant codes to the persons who areinvited to join a given conference call. The provisioning database 710responds to the VRU 705 with a signal 712 indicating the validity orinvalidity of the conference code 711 submitted by the VRU 705. If theconference code 711 submitted by the VRU 705 is invalid, the VRU 705 canre-prompt the caller 700 to reenter the conference code 711 (not shownexplicitly in FIG. 8), and can repeat this process for a reasonablenumber of times. However, if the caller 700 fails to enter a validconference code 711 within this time frame, the interaction between theVRU 705 and the caller 700 will be terminated, or the caller 700 will beforwarded to a live human operator for resolution.

3. If the conference code 711 submitted by the VRU 705 is valid, the VRU705 passes a message referencing the conference code 711 to a proxyserver 715 requesting that the caller 700 be added to a conference.Those skilled in the art can construct the proxy server 715 to realizethe functions described herein using, for example, a general-purposepersonal computer (PC) running, e.g., an operating system such as any ofthe Windows™ family of operating systems, Linux™, or Unix™, or the like.The proxy server 715 can also run application software, including atleast software relating to supporting Session Initiated Protocol(SIP)-based conferencing and/or telephony. Software suitable for thesepurposes is commercially available from, e.g., Vail Systems, Inc.(www.vailsys.com). The proxy server 715 maintains a list of all activeinterface servers 720 a, 720 b, and 720 c (referenced collectively asinterface server 720), and also tracks the loads currently beingsupported by each interface server 720. The interface servers 720function at least in part to support pipes or queues that containrequests to be acted upon by a conferencing database 725. These pipes orqueues can assume any characteristic or type known to those skilled inthe art, including but not limited to FIFO, LIFO, or other types of datastructures. The interface servers 720 can be implemented to realize thefunctions described herein using general-purpose server hardwareavailable from a variety of vendors. The proxy server 715 keeps runningcounts of pending message requests pending or awaiting on each interfaceserver 720. With this data, the proxy server 715 can perform loadbalancing across each of the various interface servers 720. Theconferencing database 725 can then select requests from various ones ofthe queues supported by the various interface servers 720 for action.The results of such action can be posted or stored in a results queue orother data structure.

4. The proxy server 715 determines that the message referencing theconference code 711 incoming from the VRU 705 should be processed by theconferencing system 735, which comprises at least the interface servers720 and the conferencing database 725. The conferencing database 725tracks state or status data for each conference currently active at anygiven time. Illustrative but non-limiting examples of the types of datagenerated or tracked by the conferencing database 725 can include, butare not limited to: a unique identifier associated with each conference,identifying each conference to the exclusion of all others; alecture-only flag as applicable to each conference; a record flagindicating whether the conference is to be or is being recorded; aunique identifier indicating the mixer 730 assigned to support eachconference; one or more conference keys associated with hosts 100 orparticipants 404 a or 404 b engaged in various conferences. For eachconference supported by the conferencing database 725, as the status ofeach conference changes, the conference database 725 tracks each changein status in a state table or other suitable data structure.

The proxy server 715 selects an available interface server 720 (either720 a, 720 b, or 720 c), and forwards the incoming message referencingthe conference code 711 to the selected interface server 720 a. FIG. 8shows three interface servers 720 a, 720 b, and 720 c for convenience inillustrating the concepts of load balancing and redundancy. However,this arrangement is not mandatory or critical to the instant invention,and various embodiments of the invention may deploy one or moreinterface servers 720, depending on the requirements and circumstancesof a given application of the instant invention.

The proxy server 715 maintains an active list of each SIP-enabled devicecurrently operative within the system shown in FIG. 8. Each SIP-enableddevice is required to register with the proxy server 715 upon start-up,and to provide thereafter “heartbeat” messages periodically (e.g., onceevery “x” seconds) to the proxy server 715. In this manner, the proxyserver 715 can determine which SIP devices are currently up and running,and further knows the current status of each SIP device. If the proxyserver 715 does not receive a “heartbeat” message from a given SIPdevice after expiration of a pre-determined time interval (e.g., onceevery “x” seconds), thus indicates to the proxy server 715 that thegiven SIP device may be inoperative. The proxy server 715 can then routefuture requests or commands to SIP devices that function as back-updevices to the apparently inoperative SIP device.

5. The conferencing system 735 then determines if a conferenceassociated with the input conference code 711 already exists, bychecking for an entry in the conferencing database 725 referencing theinput conference code 711. If the conferencing database 725 contains noreference to the input conference code 711, the conferencing system 735will request that a new conference be created, and the conferencedatabase 725 will generate a new unique conference identifier 712 forthe new conference. The conferencing system 735 will then associate thenewly generated conference identifier 712 with the conference code 711input by the VRU 705.

6. The new unique conference identifier 712 can be forwarded to the VRU705 and/or to the mixer (730 a, 730 b, or 730 c) that is selected(discussed below) to host the new conference. The VRU 705 can thenassociate all future callers 700 who provide the given conference code711 (or a code associated with the code 711) with the new uniqueconference identifier 712, and can thus use this relationship betweenthe conference code 711 and the conference ID 712 to connect thesefuture callers to the appropriate mixer 730 that is hosting theconference sought by such future callers.

7. Via a new-conference-request message or other suitable mechanism, theconferencing system 735 requests that the proxy server 715 select amixer 730 (from among e.g., the mixers 730 a, 730 b, or 730 c) to hostthe new conference, which will be associated with the newly createdunique conference identifier 712. If necessary, the conferencingdatabase 725 can stall the VRU 705 with a temporary acknowledgementcommand or other similar messaging mechanism (not shown in FIG. 8) inorder to “buy” more time to respond to the VRU's original request, i.e.,to add the caller 700 to a conference. While FIG. 8 shows this requestpassing through an interface server 720 a for convenience inillustration and discussion, in some embodiments of the invention, thismessage may pass directly to the proxy server 715 and thus bypass theinterface server 720 a.

8. The proxy server 715 selects one of the mixers 730 a, 730 b, or 730 cto host the new conference. This selection can be made based uponpresent mixer load and/or capacity (determined using any suitablealgorithm), mixer calls, or other applicable criteria. The mixers 730can be implemented using, e.g., a general-purpose server or computerincluding one or more microprocessors and any hardware necessary tosupport VoIP, or more generally, packet-based communications (e.g., oneor more specialized voice-processing boards such as those commerciallyavailable from Dialogic, a subsidiary of Intel Corporation), running asuitable operating system (e.g., Unix™, Linux™, any of the Windows™family, or the like), and running suitable application software,including at least conference mixing software available from VailSystems, Inc., as referenced above. The proxy server 715 forwards the“new conference” request to the selected mixer, e.g., mixer 730 a. Theproxy server 715 maintains data indicating the current status of each ofthe mixers 730 a, 730 b, and 730 c (collectively 730), as discussedabove with other SIP devices. Maintaining status in this manner allowsthe proxy server 715 to perform load balancing among the various mixers730, similar to the load balancing described above in connection withthe interface servers 720, and also prevents the proxy server 715 fromsending conference requests to a “dead” mixer 730. The same commentsabove directed to the illustrative number of interface servers 720 applyequally to the number of mixers 730 shown in FIG. 8.

If the selected mixer 730 a can create the new conference, it does soand returns an acknowledgement message (not shown) to the conferencingdatabase 725 via the proxy server 715. Otherwise if the selected mixer730 a cannot create the new conference, it returns a negativeacknowledgement to the conferencing database 725 via the proxy server715. In the latter instance, the proxy server 715 would then select adifferent mixer (e.g., mixer 730 b or 730 c) to host the new conference.In any event, once a suitable mixer 730 is located to host the newconference, the conferencing database 725 is updated to show that thenew conference is now assigned to the selected mixer 730 a. A uniqueidentifier corresponding to the selected mixer 730 a is stored with theconference code 711 and/or the conference identifier 712, therebyassociating the conference with the mixer 730 a. The conferencingdatabase 725 can then provide the unique conference identifier 712 tothe selected mixer 730 a and to the VRU 705, and also instructs the VRU705 to connect the caller 700 to the selected mixer 730 a.

9. In response to the command from the conferencing database 725 toconnect the caller 700 to the selected mixer 730 a, the VRU 705 sends aproposed set of IP/port data to the selected mixer 730 a (via the proxyserver 715) for routing the conference stream data (the datarepresenting the verbal or other interactions exchanged betweenconferees) between the VRU 705 and the selected mixer 730 a. The VRU 705also indicates whether the caller 700 is a conference host 100. In someembodiments of the instant invention, the conference will not actuallybegin until the host 100 associated with the conference code 711provided by the VRU 705 has dialed into the conference. Until thathappens, the various participants calling into the conference may be puton hold, but not actually bridged together into a conference until thehost calls in.

10. The selected mixer 730 a responds to the VRU 705 with the actualIP/port information that will be used for passing conferencing-relatedmedia between the mixer 730 a and the VRU 705 for communication with thecaller 700. The mixer 730 a also provides the VRU 705 with informationrelating to any media descriptions or coders-decoders (codecs) that theVRU 705 may need to process the conferencing stream as passed betweenthe caller 700 and the selected mixer 730 a, via the VRU 705. Uponreceiving this response from the selected mixer 730 a, the VRU 705configures itself to receive the conferencing stream, e.g., byactivating its RTP stream, and the caller 700 is now in the conference.“RTP” stands for real time transport protocol, which is an IETF standardfor streaming real time multimedia over an IP network in packets. Atthis point, the VRU 705 and the selected mixer 730 a are now connectedvia a local area network. Thus, each caller 700 that calls into aconference may reach the VRU 705 via a circuit-switched network (notshown), but the link between the VRU 705 and the selected mixer 730 canbe via a packet-switched (e.g., VoIP) network. The participants, 404 aand 404 b, and the host 100 can dial-in to a conference via differentports on the VRU 705, or even via different VRUs 705 altogether, butwill all be linked to a given mixer 730, e.g., mixer 730 a.

The conferencing database 725 is updated to show that the VRU 705 andthe caller 700 are in the conference, and the status of each of thevarious devices discussed herein, as well as the overall status of theconference itself, are updated with the conferencing database 725periodically.

Caller Into Existing Conference

If the caller 700 is dialing into an already-existing conference, thesame method as discussed above is performed, up to paragraph 5, wherethe conferencing database 725 checks for an existing conferenceassociated with the input conference code 711. If the conferencingdatabase 725 locates an existing conference identifier 712 correspondingto the conference code 711 submitted by the VRU 705, the conferencedatabase 725 knows that the conference sought by the caller 700currently exists. In this case, the conferencing database 725 thenforwards to the VRU 705 the identifier of the mixer 730 a that ishosting the existing conference. The VRU 705 then sends an “invite”message to the selected mixer 730 a. The selected mixer 730 a respondswith the IP/port data that the mixer 730 a will use for transmitting theconference stream media to the caller 700 via the VRU 705. The VRU 705then activates its RTP stream and the caller 700 is now in theconference. The conferencing database 725 is updated accordingly toreflect that the caller 700 is in the conference.

Message Sequence Diagrams

Having provided the above overview of the components, architecture, andmessage flow of the instant conferencing system and method in connectionwith FIG. 8, the discussion now turns to the message sequence diagramsshown in FIGS. 1-7.

This portion of the detailed description references sequence diagramsthat specify the messages sent between the VRU 705, ConferenceManagement System (CMS) 735, Mixer(s) 730, Database 725, and SIP proxyserver 715. At least the following cases are diagrammed and/or discussedherein:

 1. 1. New Conference Setup  2. 2. Additional Conferee joins an existingconference  3. 3. Conference Participant Hangs Up  4. 4. Host Hangs Up 5. 5. Host Mutes or Unmutes all participants  6. 6. Host mutes orun-mutes an individual, selected participant  7. 7. Host sets entry/exitannouncement state  8. 8. Host locks/unlocks conference  9. 9. HostDeletes an individual participant 10. 10. Host sets host-hang-upBehavior 11. 11. Host Requests Roll-Call 12. 12. Host RequestsParticipant Count 13. 13. Host Requests that Conference be Recorded 14.14. The CMS 735 Ends Conference 15. 15. Participant Drops Unexpectedly16. 16. Mixer Crashes 17. 17. Mixer Heartbeat 18. 18. RegistrationProcedures 19. 19. Outdial

The instant discussion describes illustrative embodiments ofconferencing apparatus and related methods using the known SessionInitiation Protocol (SIP). Note that the illustrative SIP messages shownhere are examples chosen for expository convenience only and do notlimit the instant invention.

1. New Conference Setup

When a host enrolls for conferencing services, he or she may specify atleast one parameter applicable to a conference involving the host. Thoseparameters that are not explicitly specified may be set to a defaultvalue by the application, based on a profile associated with the hostcode 711 used to request a particular conference. Each of the parameterscan be placed in a separate header field in the SIP INVITE message. Thehost can specify the conference parameters (e.g., those shown in Table 1below) on a per-conference basis. If a particular field is not present,a default value can be used.

TABLE 1 Parameter (Header Field Name) Possible Valuesx-on-hold-music-url A URL of a file to use for the music track. Use thestring “standard” to specify standard music. Will default to no music ifheader not present. x-periodic-message-url A URL of a file to use forthe periodic on-hold message. Use the string “standard” to play astandard message. Will default to no periodic message if header notpresent. x-upsell-message-url A URL of a file to play once to aparticipant while holding. Will default to no upsell message if headeris not present. x-entry One of: None, Name, Tone, NameAndTone. Willx-exit default to Tone if header is not present. Note they are separateheaders. x-lecture-only-flag One of: On, Off. Will default to Off ifheader is not present. x-record-flag One of: On, Off. Will default toOff if header is not present. x-max-hold-time Max # of seconds to allowparticipants to hold for this conference while waiting for the host toarrive. Value 0 means can't hold at all, host must join first. Willdefault to 10 minutes if header is not present. x-max-wrap-time Max # ofseconds to allow participants to continue on bridge after host leaves.Value of 0 indicates conference terminates immediately when host leaves.Will default to 20 minutes if header is not present.

The various message flows shown in the various drawing figures arenumbered for ease of reference, and these flows are discussed herein.

Turning to FIG. 1:

1. The VRU 705 receives a call from the caller. In response to promptsfrom the VRU 705, the caller 700 enters his or her PIN (user-specifiedor system-specified), other optional information relating to conferenceconfiguration, and the conference code (e.g., 577342), which can beeither a host code or a participant code. The VRU 705 sends an SIPINVITE message to the proxy server 715. In FIG. 1, the proxy server 715appears twice, but this representation is only to avoid over-crowdingand confusion in the message sequence diagrams. The two blocks labeled715 can represent one proxy server 715.

This INVITE message may take the following illustrative but non-limitingform:

INVITE sip:proxy1.string1.com SIP/2.0

Via: SIP/2.0/UDP eos2002.string1.com

From: sip:sipdaemon@eos2002.string1.com

To: sip:orchestra@string1.com

Call-ID: 2002331059963@eos2002.string1.com

Content-Length: 307

Content-Type: application/sdp

x-host-code: 577342

x-participant-code: 577343 (looked up by the VRU)

x-max-hold-time: 1200

x-max_wrap-time: 0

x-entry: NameAndTone

x-participant-type: Host

x-client: att

x-site: Denver

x-email-notify: Y

x-email-address: bigguy@company.com

x-phone: 3321147

String 1 represents a string that can, for example identify a particularclient or company deploying the instant invention, and may comprise atleast in part, a URL address.

The invite message can contain data related to specifying parametersrelated to the known Session Description Protocol (SDP), such theconferencing media type (e.g., audio), parameters relating to anycoders/decoders (codecs) used in connection with the media, and one ormore transport addresses related to the conference.

2. The Proxy Server 715 determines that this message is for theconference management system (CMS) 735, selects an available interfaceserver 720, and forwards the INVITE message to the selected interfaceserver 720. Assume that the interface 720 a shown in FIG. 8 is chosen.This INVITE message as forwarded to the interface server 720 a can takethe following illustrative form:

INVITE sip:orchestra1.string1.com SIP/2.0

Via: sip:proxy1.string1.com

Via: SIP/2.0/UDP eos2002.string1.com

From: sip:sipdaemon@eos2002.string1.com

To: sip:CMS@string1.com

Call-ID: 2002331059963@eos2002.string1.com

Content-Length: 6207

Content-Type: application/sdp

x-host-code: 577342

x-max-hold-time: 1200

x-max_wrap-time: 0

x-entry: NameAndTone

x-participant-type: Host

x-client: att

x-site: Denver

x-email-notify: Y

x-email-address: bigguy@company.com

x-phone: 3321147

3. The CMS 735 accesses the conferencing database 725 to determinewhether a conference corresponding to the input conference code 711already exists, or should now be created. In this case, assume a newconference should be created. The CMS 735 determines it needs to createa new conference, and based on the input conference code 711, it alsodetermines that it needs to use e.g., a “Class A” mixer to host the newconference. Various classes of mixers are discussed below. The CMS 735also generates a globally unique conference string that identifies thisnew conference. For this example, assume the CMS 735 generates theconference string “3xyza1077fkdda”. This string will be passed to thechosen mixer 720 and to the VRU 705 to identify the particularconference to which the caller 700 (i.e., host 100) should be joined.Subsequent callers 700 who enter a participant code or conference code711 that corresponds to the same conference will be associated with thesame unique conference identifier. Since the CMS 735 will have tocontact the mixer 730 to create the new conference, which might taketime, the CMS 735 determines that it may need to send a TRYING responseto the VRU 705.

4. The CMS 735 sends back to the VRU 705 a 100 Trying message, which isconsidered a response to the INVITE request sent previously by the VRU705 in step 1 above. While not strictly necessary, it may be preferableto do this as it buys more time for the CMS 735 to perform its taskwithout the VRU 705 thinking that the CMS 735 has become inoperative. Inthis step, the response can go from the CMS 735 to the proxy server 715.The TRYING 100 message can take the following illustrative form:

SIP/2.0 100 TRYING

Via: sip:proxy1.string1.com

Via: SIP/2.0/UDP eos2002.string1.com

From: sip:sipdaemon@eos2002.string1.com

To: sip:CMS@string1.com

Call-ID: 2002331059963@eos2002.string1.com

5. The proxy server 715 can forward the TRYING response to the VRU 705to “stall” the VRU 705, using the following illustrative form:

SIP/2.0 100 TRYING

Via: SIP/2.0/UDP eos2002.string1.com

From: sip:sipdaemon@eos2002.string1.com

To: sip:CMS@string1.com

Call-ID: 2002331059963@eos2002.string1.com

6. To request creation of the new conference, the CMS 735 can send thefollowing SIP message to the proxy server 715, which can in turn forwardthis message to one or more mixers 730. This message can take thefollowing form:

INFO sip:proxy1.string1.com SIP/2.0

Via: SIP/2.0/UDP CMS.string1.com

From: sip:CMS@string1.com

To: sip:mixerA@string1.com

Call-ID: 2002331059963@CMS1.string1.com

x-command: CreateNewConference

x-conference-id: 3xyza1077fkdda

x-entry: NameAndTone

x-exit: Tone

x-lecture-only-flag: N

x-record-flag: N

x-participant-type: Host

7. The proxy server 715 receives the message sent in the previous step,and in response thereto, selects a mixer 730 to hose the new conference.Assuming the new conference should be hosted on a “type A” mixer 730,the proxy server 715 searches for a mixer 730 that registered as, e.g.,a mixer of type “mixer-A” that is least loaded and forwards the INFOmessage as shown below. In this case, assume that the proxy server 715selected a mixer 730 a identified by the string “mixer13.string1.com” asthe least loaded mixer of the type “mixer-A”, and sends the selectedmixer 730 a a message taking the following illustrative form:

INFO sip:mixer13.string1.com SIP/2.0

Via: SIP/2.0/UDP proxy1.string1.com

Via: SIP/2.0/UDP CMS1.string1.com

From: sip:CMS@string1.com

To: sip:mixerA@mixer13.string1.com

Call-ID: 2002331059963@CMS1.string1.com

x-command: CreateNewConference

x-conference-id: 3xyza1077fkdda

x-entry: NameAndTone

x-exit: Tone

x-lecture-only-flag: N

x-record-flag: N

x-participant-type: Host

8. The selected mixer 730 a sees from the INFO message above that a newconference is to be created using conference identifier“3xyza1077fkdda”. The mixer 730 a can then establish the new conferenceinternally and initialize it based on the parameters specified in theINFO message. It can then send a 200 OK message to the proxy server 715in the following illustrative form:

SIP/2.0 200 OK

Via: SIP/2.0/UDP proxy1.string1.com

Via: SIP/2.0/UDP CMS1.string1.com

To: sip:mixerA@mixer13.string1.com

From: sip:CMS@string1.com

Call-ID: 2002331059963@CMS.string1.com

x-command: CreateNewConference

x-conference-id: 3xyza1077fkdda

x-max-hold-time: 1200

x-max_wrap-time: 0

x-entry: NameAndTone

The selected mixer 730 a can use the above 200 OK response to indicatethat it accepts the request for a new conference. If for any reason itcould not establish the new conference, it would use one of the SIPfailure response codes (for example, 480 or 486).

9. The INFO response message from the previous step is forwarded by theproxy server 715 back to the CMS 735.

10. The CMS 735 updates the conferencing database 725 to reflect that anew conference associated with the unique conference identifier“3xyza1077fkdda” is hosted on the selected mixer 730 a, which isidentified by the unique mixer string.

11. After the CMS 735 updates the conferencing database 725, it sends a302 response message to the VRU 705 (via the proxy server 715), takingthe following illustrative form:

SIP/2.0 302 Moved Temporarily

Via: sip:proxy1.string1.com

Via: SIP/2.0/UDP eos2002.string1.com

To: sip:CMS@CMS1.string1.com

From: sip:sdpdaemon@eos2002.string1.com

Call-ID: 2002331059963@eos2002.string1.com

Contact: sip:mixer13.string1.com

x-conference-id: 3xyza1077fkdda

x-max-hold-time: 1200

x-max_wrap-time: 0

x-entry: NameAndTone

x-exit: Tone

x-participant-type: Host

x-attendee-id: 1

At this point, the VRU 705 knows that it should send the INVITE messageto the selected mixer 730 a and that it should use the conferenceidentifier string (conference-id) “3xyza1077fkdda” andattendee-identifier (attendee-id) “1” in connection with the givenconference and the given conferee, respectively. The conference-id andattendee-id are generated by the CMS 735 and sent back to the mixer 730in the above 302 response. The VRU 705 copies these new headers into theINVITE message that it sends directly to the selected mixer 730 a instep 15 below.

12. The Moved Temporarily 302 message from the previous step isforwarded from the proxy server 715 to the VRU 705.

13. The VRU 705 acknowledges (ACKS) the INVITE response that the CMS 735sent.

14. The ACK request is forwarded from the proxy server 715 to the CMS735.

15. The VRU 705 sends an INVITE message to the selected mixer 730 a, viathe proxy server 715. This INVITE message contains the IP/portinformation proposed by the VRU 705 for the RTP/RTCP stream that willcomprise the conferencing stream going to and from the caller 700. Also,note that the INVITE message that the VRU 705 sends to the mixer 730contains the header x-string1-participant-type. Preferably, this headerspecifies whether the caller 700 is a host 100. In some embodiments ofthe instant invention, if this header is absent, the mixer 730 willassume that the incoming caller 700 is not the host. In theseembodiments, the mixer 730 needs to know when the host 100 arrives,because if participants call in before the host 100, they will be put onhold but not bridged together. Other alternative embodiments forhandling a scenario wherein the participants call in before the host 100are discussed below. The INVITE message can take the followingillustrative form:

INVITE sip:proxy1.string1.com SIP/2.0

Via: SIP/2.0/UDP eos2002.string1.com

From: sip:sipdaemon@eos2002.string1.com

To: mixerA@mixer13.string1.com

CALL-ID: 2002331059963@eos2002.string1.com

Content-Length: 307

Content-Type: application/sdp

x-Conference-id: 3xyza1077fkdda

x-entry: NameAndTone

x-exit: Tone

x-participant-type: Host

x-max-hold-time: 1200

x-max_wrap-time: 0

x-attendee-id: 1

The entity body of this message can contain the transport address, mediatype, and codec parameters in the SDP.

16. The proxy server 715 receives the INVITE message sent in theprevious steps and forwards it to the selected mixer 730 a.

17. The mixer 730 a receives the INVITE message sent in the previousstep, and sends a response to that message back to the proxy server 715.This time, the SDP in the entity body contains data representing the IPaddress and port that the mixer 730 will use for the RTP/RTCP datacomprising the conferencing voice stream, along with the description ofthe media (codec) that the mixer 730 will send back to the VRU 705. Theresponse sent by the mixer 730 back to the proxy server 715 can take thefollowing illustrative form:

SIP/2.0 200 OK

Via: SIP/2.0/UDP proxy1.string1.com

Via: SIP/2.0/UDP eos2002.string1.com

To: sip:mixerA@mixer13.string1.com

From: sip:sipdaemon@eos2002.string1.com

CALL-ID: 2002331059963@eos2002.string1.com

Content-Length: 262

Content-Type: application/sdp

x-conference-id: 3xyza1077fkdda

x-Max-Hold-Time: 1200

x-Max_Wrap-Time: 0

x-entry: NameAndTone

x-exit: Tone

x-attendee-id: 1

x-participant-type: Host

18. The proxy server 715 forwards the response to the VRU 705. Once theVRU 705 activates its RTP stream the caller 700 is in the conference.

19. The VRU 705 ACKS the INVITE response received in the previous step,thereby confirming that the VRU 705 is in the conference.

20. The proxy server 715 forwards the ACK to the selected mixer 730 a.

21-28. Now that the caller 700 has successfully been added to theconference, the conferencing database 725 is updated. Whenever the stateof any conference between any VRU 705 and any mixer 730 changes, a SIPINFO message is sent to the CMS 735 so that the conferencing database725 is updated on of the state of every conference. Thus, since thecaller 700 has been added to the conference, the SIP INFO message issent to the proxy server 715, which in turn will forward it to the CMS735. If the VRU 705 had been unable to join the conference for somereason (e.g., no response or an error response from the mixer 730), itwould inform the CMS 735 of this condition with a different INFOmessage. The SIP INFO message sent to the CMS 735 to update theconference status with the conferencing database 725 can take thefollowing illustrative form:

INFO proxy1.string1.com SIP/2.0

Via: SIP/2.0/UDP eos2002.string1.com

From: sip:sipdaemon@eos2002.string1.com

To: sip: CMS@string1.com

Call-ID: 2002331059963@eos2002.string1.com

x-command: StatusUpdate

x-conference-id: 3xyza1077fkdda

x-status-change: JoinedConference

Note that both the VRU 705 and the mixer 730 can send INFO messages tothe CMS 735. In each case, the INFO message can go through the proxyserver 715, and, in each case, the CMS 735 can respond to the INFOmessage to confirm receipt.

29. The CMS 735 updates the conferencing database 725 to indicate thatthe caller 700, who is communicating through a given port of the VRU705, is now in the conference.

2. Additional Conferee Joins an Existing Conference

Turning to FIG. 2:

1. The caller 700 dials into the VRU 705, is authorized/authenticated bythe VRU 705, and then enters his/her participant ID or conference key(collectively referenced as 711), which can be identical to or otherwiserelated to the conference key 711 supplied by the previous caller 700,as discussed in connection with FIG. 7. The VRU 705 sends an SIP INVITEmessage to the proxy server 715 taking the following general form:

INVITE sip:proxy1.string1.com SIP/2.0

Via: SIP/2.0/UDP eos2002.string1.com

From: sip:sipdaemon@eos2002.string1.com

To: sip:CMS@string1.com

Call-ID: 2002391069063@eos2002.string1.com

Content-Length: 6207

Content-Type: application/sdp

x-host-code: 577342

2. The proxy server 715 forwards the above INVITE message and theconference key 711 input by the caller 700 in the previous step to aninterface server 720 in the CMS 735. The interface server 720 can beselected as discussed above in connection with FIG. 1.

3. The CMS 735 receives the INVITE message sent in the previous step andthe searches for the input conference key or code 711 in theconferencing database 725. The conferencing database 725 preferablycontains an entry for the active conference corresponding to the inputconference key, e.g., 577342. In the instant discussion, assume that thecaller 700 discussed in connection with FIG. 2 wishes to join the newconference created in connection with FIG. 1. Thus, the CMS 735 willrecognize that the conference sought by the caller 700 is an existingconference. The CMS 735 will further recognize that the uniqueconference identifier string for this existing conference is“3xyza1077fkdda”, and that this existing conference is being hosted onthe mixer 730 associated with the unique mixer identifier string“mixer13.string1.com”.

4. Based on the information obtained in the previous step, the CMS 735sends (via the proxy server 715) an INVITE response to the VRU 705 inthe following illustrative form:

SIP/2.0 302 Moved Temporarily

Via: SIP/2.0/UDP eos2002.string1.com

To: sip:CMS@CMS1.string1.com

From: sip:sdpdaemon@eos2002.string1.com

Call-ID: 2002391069063@eos2002.string1.com

Contact: mixer13.string1.com

x-conference-id: 3xyza1077fkdda

5. The proxy server 715 forwards the INVITE response from the previousstep to the VRU 705.

6. The VRU 705 receives the INVITE response sent in the previous step,and sends an acknowledgement (ACK) to the CMS 735 as a reply to theINVITE response.

7. The proxy server 715 forwards the ACK to the CMS 735.

8. The VRU 715 sends a new INVITE message, this time destined for theselected mixer 730 a that is hosting the existing conference. The INVITEmessage can be transmitted via the proxy server 715, and can take thefollowing illustrative form:

INVITE sip:proxy1.string1.com SIP/2.0

Via: SIP/2.0/UDP eos2002.string1.com

From: sip:sipdaemon@eos2002.string1.com

To: mixerA@mixer13.string1.com

Call-ID: 2002391069063@eos2002.string1.com

Content-Length: 6207

Content-Type: application/sdp

x-conference-id: 3xyza1077fkdda

9. The proxy server 715 forwards the INVITE message to the selectedmixer 730 a.

10. The selected mixer 730 a sends a response for the INVITE messageback to the proxy server 715. This time, the transport address containedin the entity body (SDP) contains the IP and port that the mixer 730will use for the RTP/RTCP data passing between the selected mixer 730 aand the caller 700 via the VRU 705, once the caller is connected to theconference. This response can take the following illustrative form:

SIP/2.0 200 OK

Via: SIP/2.0/UDP eos2002.string1.com

To: sip:mixerA@mixer13.string1.com

From: sip:sipdaemon@eos2002.string1.com

Call-ID: 2002391069063@eos2002.string1.com

x-conference-id: 3xyza1077fkdda

Content-Length: 380

Content-Type: application/sdp

11. The proxy server 715 forwards the message from the previous step tothe VRU 705.

12. The VRU 705 sends an ACK request to the mixer 730 to acknowledge theINVITE Response sent by the mixer 730 a in step 10.

13. The proxy server 715 forwards the ACK message to the selected mixer730 a.

14-21. Once the VRU 705 activates its RTP stream, the new caller 700 isin the conference. The VRU 705 sends an INFO message (via the proxyserver 715) to the CMS 735 to inform the CMS 735 that the caller 700 wassuccessfully added to the conference.

INFO proxy1.string1.com SIP/2.0

Via: SIP/2.0/UDP eos2002.string1.com

From: sip:sipdaemon@eos2002.string1.com

To: sip: CMS@string1.com

Call-ID: 2002391069063@eos2002.string1.com

x-conference-id: 3xyza1077fkdda

x-status-update: JoinedConference

A similar INFO message is also sent from the selected mixer 730 to theCMS 735, and all the INFO messages are responded to as shown in thediagram.

22. The CMS 735 updates the conferencing database 725 to indicate thatthe additional conferee, the second caller 700, is now in the givenconference.

3. Conference Participant Hangs Up

Turning to FIG. 3:

1. A given conferee, in this case a conference participant as opposed tothe conference host, hangs up his or her handset, sending a disconnectsignal to the VRU 705. Alternatively, the conferee can enter some DTMFsignals instructing the conferencing application to disconnect or hangup on his/her behalf. The VRU 705 sends a BYE message to the mixer 730hosting the given conference. Note that the BYE message can go directlyto the mixer 730 without passing through a proxy server 715. The BYEmessage can take the following illustrative form:

BYE sip:mixerA@mixer13.string1.com SIP/2.0

From: sip:sipdaemon@eos2002.string1.com

To: mixerA@mixer13.string1.com

Call-ID: 2002391069063@eos2002.string1.com

x-conference-id: 3xyza1077fkdda

The BYE message includes a caller identifier (Call-ID) that identifiesthe particular caller 700 after that caller 700 has been admitted to agiven conference. Note that the Call-ID parameter in the BYE messagepreferably matches the one referenced in the original INVITE messagesent by the VRU 705 to the mixer 730 when that caller 700 entered theconference. This convention enables the mixer 730 to identify exactlywhich conferee is leaving the conference and can enable theidentification of the conferee as a participant rather than a host(e.g., by the database 725).

2. The mixer 730 hosting the conference removes the participant from theconference and sends the BYE Response message back to the VRU 705through which the participant was communicating with the conference.Again, this message can be direct, i.e., without passing through theproxy server 715. This BYE message can take the following illustrativeform:

SIP/2.0 200 OK

To: sip:mixerA@mixer13.string1.com

From: sip:sipdaemon@eos2002.string1.com

Call-ID: 2002391069063@eos2002.string1.com

3-10. The VRU 705 to which the given participant was connected sends anINFO message to the CMS 735 to indicate that the given participant hasdropped from the conference. This message can take the followingillustrative form:

INFO sip:proxy1.string1.com SIP/2.0

From: sip:sipdaemon@eos2002.string1.com

To: sip:CMS@string1.com

Call-ID: 2002391069063@eos2002.string1.com

x-conference-id: 3xyza1077fkdda

x-status-update: LeftConference

The mixer 730 hosting the conference from which the participant droppedcan also send an INFO message to the CMS 735 to indicate that the givenparticipant has dropped from the conference. This message can take thefollowing illustrative form:

INFO sip:proxy1.string1.com SIP/2.0

From: sip:mixerA@mixer13.string1.com

To: sip:CMS@string1.com

Call-ID: 2002391069063@eos2002.string1.com

x-conference-id: 3xyza1077fkdda

x-status-update: LeftConference

These INFO messages can go through the proxy server 715, and can beresponded to by the CMS 735 with INFO Response messages, as shown inFIG. 3.

11. The CMS 735 updates the conferencing database 725 to indicate thenew state of the conference after the given participant has disconnectedor hung up.

4. Host Hangs Up

Turning to FIG. 4:

1. The host (as opposed to one of the participants) hangs up his/herhandset, sending a disconnect signal to the VRU 705 to which the hostis/was connected. The VRU 705 sends a BYE message directly to the mixer730 hosting the conference from which the host disconnected.

2. The mixer 730 removes the host and responds with a BYE Responsemessage, sent directly to the VRU 705.

3-10. The VRU 705 sends an INFO message to the CMS 735 via the proxyserver 715 informing the CMS 735 that the participant is out of theconference. At this point, the VRU 705 or the mixer 730 may notrecognize whether the conferee who disconnected is a host or aparticipant. The mixer 730 also sends a similar message to the CMS 735.The CMS 735 responds via the proxy server 715 to each INFO message fromthe VRU 705 and the mixer 730.

11. The CMS 735 checks the conferencing database 725 and sees that theparticipant that just left the conference was the host, and seesfurthermore that the conference is configured to end when the hostleaves.

12. The CMS 735 sends an INFO message to the mixer 730 hosting theconference with a command to terminate the conference (e.g., x-Command:TerminateConference). The CMS 735 can wait a small amount of time, e.g.5 seconds, before sending this message. This would allow non-hostparticipants to say good-bye, etc., after the host hangs up, instead ofbeing abruptly disconnected. A header (e.g., x-conference-id) ispreferably included to specify which conference to terminate.

13. The mixer 730 responds with an INFO response message.

14. The mixer 730 goes through the list of all conferees (participants)remaining in the conference and sends each a BYE message to eachconferee, via the VRU 705 ports through which one or more participantsmay be connected to the conference.

15. The VRU 705, upon receipt of the BYE message from the mixer 730,sends a BYE response back to the mixer 730.

16-23. The VRU 705 b and the mixer 730 both send INFO messages (via theproxy server 715) to the CMS 735 to inform the CMS 735 that the variousparticipants have left the conference.

24. The CMS 735 updates the conferencing database 725 to reflect thatthe various participants have been removed from the conference.

Note that steps 14 through 24 would be repeated for each remainingconferee in the conference. Note also that FIG. 4 includes two VRUs 705a and 705 b to illustrate that different participants can access a givenconference via different VRUs.

5. Host Mutes or Unmutes all Participants

The VRU 705 can send a request to mute all conference participants tothe CMS 735 instead of directly to the mixer 730 that is hosting theconference. Either way may suffice, however, the host might execute thiscommand via a web interface to the CMS 735, in which case the commandwould most likely go to the CMS 735 first. This way, the message flow isthe same for either DTMF input or web input of the command once themessage gets to CMS 735, if this consistency in message flow isimportant in a given application of the instant teachings.

Turning to FIG. 5:

1. The conference host (as opposed to a participant) enters DTMF signalsindicating he/she wants to mute or un-mute all participants. The VRU 705a (assuming the host is coupled to this VRU) sends a SIP INFO request tothe CMS 735 via the proxy server 715. The INFO request contains aspecial header (e.g., X-Command: Mute-All) and also contains the headerx-conference-id to indicate the conference to which the mute commandapplies. Also, as usual, the INFO message has a Call-ID header thatidentifies the unit, line, and setup time of the sender, who is theconference host in this instance.

2. The INFO message from the previous step is forwarded to the CMS 735.Alternatively, if the host enters the command via the web interface tothe CMS 735, the message flow may start at this point.

3. The CMS 735 checks the conferencing database 725 to see if therequest is valid (e.g., does the conference exist, is the sender reallythe host?). Assuming this request is valid, the CMS 735 updates theconferencing database 725 to indicate that this conference is marked asmuted or un-muted. The CMS 735 will send an INFO Response message backto the VRU to indicate that the (un)mute command succeeded in thefollowing steps. If the request had been invalid, the INFO Responsewould indicate failure.

4. The CMS 735 sends an INFO message to the mixer 730 that is hostingthe conference for which the Mute-All or Un-mute All command was issued.This INFO message contains the header x-Command: Mute-All (Or x-Command:Un-mute-Ali) along with the x-conference-id header.

5. The mixer 730 mutes or un-mutes all participants in the specifiedconference, and sends the INFO Response message back to the CMS 735. TheINFO Response will have a 200 OK status code, and the other headers willbe echoed from the request.

6. The CMS 735 sends the Info Response message back to the VRU 705 a.Assuming that the request was valid, the response code will be 200 OK.This step can be skipped if the mute/un-mute command originated via theweb interface to the CMS 735.

7. The Info Response message is forwarded by the proxy server 715 to theVRU 705 a.

8. The CMS 735 updates the conferencing database 725 to indicate thatthe participants are muted or unmated in response to the host's command.Note that at this point the participants may not realize that they aremuted—the VRUs 705 a and/or 705 b in the conference may still be sendingRTP streams representing speech or voice originating with theparticipants. In the case of an un-mute command, the VRUs 705 a and/or705 b may not yet be sending RTP streams. The following steps addressthis scenario.

9. The mixer 730 sends an INVITE message to the VRU (e.g., 705 b,although some participants may be coupled to the VRU 705 a, or to otherVRUs 705) for each participant in the conference. The INVITE messagecontains updated an SDP in the entity body to indicate that theparticipant is to be muted or un-muted. Also, the Call-ID header willreflect the unit, line, and setup time of the participant to be muted orun-muted. Note that the VRU 705 b may need to know who is being muted orun-muted, as that VRU 705 b may be connected to many participants inthis and other conferences.

10. The VRU 705 b sends the INFO Response message to the mixer 730 toconfirm the previous command.

11-14. The VRU 705 b sends an INFO message to the CMS 735, via the proxyserver 715, to inform the CMS 735 that it has muted or un-muted theappropriate ports to which the indicated participants are connected, asinstructed. This INFO message can use the Call-ID header to specify theparticipant and the x-StatusUpdate: Muted header can indicate that theparticipant has been muted (i.e., the participant's VRU port has stoppedsending RTP packets to the mixer 730), or x-west-StatusUpdate: Un-mutedto indicate that the RTP stream has resumed in the case of an un-mutecommand.

15. The CMS 735 updates the conferencing database 725 to reflect the newstate of the participants and/or the conference as a whole as beingeither muted or un-muted.

6. Host Mutes or Un-Mutes an Individual, Selected Participant

When the host wants to mute or un-mute an individual, selectedparticipant, the participant can be specified by the order in which theyjoined the conference. The first person joining the conference can bedesignated ‘1’, the second person designated ‘2’, and so on. The hostmay or may not readily know the order in which conferees joined theconference, depending on the specific conferencing application. If audioannouncements are enabled, the host may request and listen to the rollcall to determine the order, because the mixer 730 can be configured toplay the roll call in the order in which the participants joined, alongwith possibly a voice sample of the participant speaking his/her namewhen entering the conference. Also, the web interface to the CMS 735 canprovide information to the host about the order, e.g., by displaying theANI of each participant and the order in which they joined theconference. The ANI could be mapped to a corresponding name using one ormore databases storing such information. Suitable databases may beavailable from vendors such as Targusinfo (www.targusinfo.com).

Referring back to FIG. 5:

1. The host enters DTMF signals indicating he/she wants to mute orun-mute participant number k, where k is an integer between 1 and thenumber of participants in the conference with the participant to bemuted being identified via, e.g., a roll call of participants, asdiscussed above. The VRU 705 a sends an SIP INFO request to the CMS 735via the proxy server 715. The INFO request can contain a special headerin the general form of x-Command: Mute k and also can contain the headerx-conference-id to indicate the conference to which the command applies.Also, as usual, the INFO message can include a Call-ID header thatidentifies the unit, line, and setup time of the sender from whom themute or un-mute command originates.

2. The INFO message from the previous step is forwarded to the CMS 735.Alternatively, if the host enters the command via the web interface tothe CMS 735, the message flow would start at this point.

3. The CMS 735 checks the conferencing database 725 to see if therequest is valid (e.g., does the conference exist, is the sender reallythe host, etc.). Assuming the request is valid, the CMS 735 updates theconferencing database 725 to indicate that participant k is marked asmuted or un-muted. The CMS 735 then sends the INFO Response message backto the VRU 705 a to indicate that the mute or un-mute command succeededin the following steps. If the request had been invalid, the INFOResponse would indicate failure.

4. The CMS 735 sends an INFO message to the mixer 730 that is hostingthe conference for which the mute or un-mute command was issued. ThisINFO message can contain a header in the general form of x-Command: Muteor x-Command: Un-mute along with the x-conference-id header. Also thex-attendee-id header can be included to identify which conferenceparticipant is being muted or un-muted.

5. The mixer 730 mutes or un-mutes the specified participant and sendsthe INFO Response message back to the CMS 735. The INFO Response willhave a 200 OK status code, along with the other headers echoed from theoriginal request.

6. The CMS 735 sends the Info Response message back to the proxy server715. Assuming the request was valid, the response code will be 200 OK.This step can be skipped if the command originated via the webinterface.

7. The Info Response message is forwarded by the proxy server 715 to theVRU.

8. The CMS 735 updates the conferencing database 735 to indicate thatthe participant is muted or un-muted.

9. The mixer 730 sends an INVITE message to the VRU 705 through whichthe now muted or un-muted participant communicates with the conference.The INVITE message contains updated SDP data to indicate that theparticipant is now muted or unmated in response to the input command.Also the Call-ID header will reflect the unit, line, and setup time ofthe participant to be muted or un-muted.

10. The VRU 705 b sends the INVITE Response message to the mixer 730 toconfirm the previous command.

11-14. The VRU 705 b sends an INFO message to the CMS 735, via the proxyserver 715, to inform the CMS 735 that it has muted or un-muted itself.This INFO message can use the Call-ID header to specify the participant.The INFO message can use a header in the general form of thex-status-update: Muted to indicate that the participant has been muted(i.e., the VRU 705 b has stopped sending any RTP packets containingvoice that originated with the muted participant), or can use a headerin the general form of x-status-update: Un-muted to indicate that theparticipant has been un-muted (i.e., the VRU 705 b has resumed sendingRTP packets containing voice that originated with the un-mutedparticipant.

15. The CMS 735 updates the conferencing database 725 to reflect the newstate (either muted or un-muted) of the participant.

7. Host Sets Entry/Exit Announcement State

At least four illustrative states for entry/exit announcements arediscussed herein: off, tones only, name only, and both name and tones.

Turning to FIG. 6:

1. The host can enter predefined DTMF signals to set the entry exitstate to tones only. The VRU 705 to which the host is coupled sends anINFO message to the CMS 735, via the proxy server 715. The INFO messagecan contain a header in the general form of x-command: SetEntryExitTone. The message can also contain the x-conference-id header and theCall-ID header.

2. The proxy server 715 forwards the INFO message to the CMS 735.

3. The CMS 735 checks the conferencing database 725 to see if therequest is valid (e.g. does the conference exist, is the sender reallythe host, etc.). Assuming the request is valid, the CMS 735 updates theconferencing database 725 to indicate the new entry/exit announcementstatus for the conference. The CMS 735 will send an INFO Responsemessage back to the VRU 705 to indicate that the command succeeded inthe following steps. If the request had been invalid, the INFO Responsewould indicate failure.

4. Assuming the entry/exit state is to be changed as requested, the CMS735 sends an INFO message to the mixer 730 that is hosting the specifiedconference. This INFO message can contain a header in the general formof x-command: set-entry-exit Tone, and can also contain thex-conference-id header.

5. The mixer 730 changes the entry/exit announcement state internally,and sends an INFO Response back to the CMS 735 to indicate success.

6. The CMS 735 sends an INFO Response back to the VRU 705, via the proxyserver 715.

7. The proxy server 715 forwards the INFO Response to the VRU 705.

8. The CMS 735 updates the conferencing database 725 to reflect thechanges in entry/exit announcement state as necessary.

8. Host Locks/Unlocks Conference

The host can lock a conference, which means that no new participants canjoin the conference until the conference is unlocked. Since the CMS 735brokers all requests to join the conference, it can handle the signalingrelated to the lock command, and the mixer 730 hosting the conferencemay not need to track or know if a conference is locked or unlocked.

Turning to FIG. 7:

1. The host enters predefined DTMF signals to lock or unlock theconference. The VRU 705 to which the host is coupled sends an INFOmessage to the CMS 735 via the proxy server 715. The INFO message cancontain a header in the general form of x-command: LockConference orx-command: UnlockConference, as the case may be. The INFO message canalso contain the x-conference-id header and the Call-ID header.

2. The proxy server 715 forwards the INFO message to the CMS 735.

3. The CMS 735 checks the conferencing database 725 to validate therequest to lock/unlock the conference (e.g., does the conference exist,is the sender really the conference host, etc).

4. The CMS 735 changes the state of the specified conference to lockedor unlocked, depending on the command issued. If participants try tojoin a locked conference, they will be rejected, put on hold, notifiedwith an appropriate message, or the like. Should additional participantsrequest access to the locked conference, a VRU 705 coupled to suchadditional participant will send an INVITE or equivalent message to theCMS 735. The CMS 735 would see that the conference is locked, and wouldsend an INVITE Response code of 480 (Temporarily not available), oranother failure response code to the VRU 705.

In other embodiments of the invention, the lock/unlock functionalitycould be handled only by the VRU 705 by accessing the conferencingdatabase 725 directly, essentially bypassing the CMS 735. This directread or access of the conferencing database by the VRU 705 isrepresented by the dashed line 777 shown in FIG. 7. When a new potentialparticipant calls in, the VRU 705 would determine whether the conferencewas locked by directly reading or accessing the conferencing database725. In further embodiments of the invention, some combination of thetwo approaches could be taken: the VRU 705 could check the conferencingdatabase 725, but if the VRU 705 does not check, the CMS 735 can stillcheck to ensure that no new participants are added to a lockedconference.

5. The CMS 735 sends an INFO Response back to the VRU 705 via the proxyserver 715 indicating a success status such as 200 OK.

9. Host Deletes an Individual Participant

It is important to manage carefully the deletion of participants toensure participants are not deleted accidentally or by mistake. Oneapproach is to have the host delete a participant based on theparticipant's attendee ID. The CMS 735 assigns each participant in theconference a attendee ID that is unique at least within the conference(and perhaps globally unique as well) upon joining a given conference.The CMS 735 ensures that an assigned attendee ID is unique at leastwithin a given conference. The host of the conference can determine theattendee ID of a conference participant by listening to a roll call orby using the web interface to the CMS 735. To facilitate quick rollcalls, the CMS 735 preferably uses relatively truncated identifiers forthe attendee ID, e.g., start counting at “1”. These identifiers can takeinteger, character, or other form as recognized by those skilled in theart. When a participant is deleted or hangs up, the attendee ID is nolonger used for that conference for any conferees joining the conferencethereafter, thereby preventing subsequent confusion that could causeerroneous deletion of a participant.

1. The host enters predefined DTMF signals to request that a participantwith an attendee-ID k, where k is an integer ranging from 1 to thenumber of participants in a given conference, be removed from theconference. The VRU sends an INFO Remove message to the CMS 735 via theproxy server 715 with a header in the general form of x-command:RemoveParticipant. Also included can be the x-conference-id header andthe x-attendee-id header to specify the attendee ID of the participantto remove.

2. The proxy server 715 forwards the INFO message to the CMS 735.

3. The CMS 735 checks the conferencing database 725 to validate therequest. (e.g., conference exists, sender is host, etc.).

4. The CMS 735 sends an INFO message to the mixer 730, echoing the INFORemove command sent from the VRU 705 in step 1 above.

5. The mixer 730 sends an INFO Response message back to the CMS 735. Themixer 730 then removes the participant specified in the command from theconference, for example, by disabling or terminating the link betweenthe (either physical or logical) mixer 730 and the VRU 705 over whichthe deleted participant formerly communicates.

6. The CMS 735 sends INFO response back to the VRU 705 via the proxyserver 715.

7. The mixer 730 sends a BYE message directly to the VRU 705 to whichthe deleted participant is coupled (which may or may not be the same VRU705 to which the conference host is coupled) to inform the deletedparticipant that he/she has been deleted.

8. The VRU 705 responds to the BYE message and sends the same to themixer 730.

9. The VRU 705 sends an INFO message to the CMS 735 (via the proxyserver 715). The message contains a header in the general form ofx-status-update: LeftConference, along with the Call-ID field toindicate the participant to whom the message refers.

10. The conferencing database 725 is updated to indicate that theconference no longer includes the deleted participant.

10. Host Sets Host-Hang-Up Behavior

The behavior of the conferencing application when the host hangs up cantake one of at least two states, which are selectable by the host. Theconference can terminate when the host hangs up, or it can continueuntil all participants have hung up (or all except 1). The x-commandheader can be used in connection with this command. An illustrativeembodiment of this command can take the general form of x-command:SetTerminateOnHostHangup [True|False]. An illustrative sequence ofmessages to implement this command can include at least the following:

1. The host enters predefined DTMF signals to set the conferencebehavior to terminate when the host hangs up. The VRU 705 to which thehost is coupled sends an INFO message to the CMS 735 via the proxyserver 715. The INFO message can contain a header in the general form ofx-Command: SetTerminateOnHostHangup True, and also a header in the formof x-conference-id.

2. The proxy server 715 forwards the above INFO message to the CMS 735.

3. The CMS 735 checks the conferencing database 725 to validate therequest, illustrative examples of which are discussed above. If therequest is valid, the CMS 735 updates the conferencing database 725 toreflect the new host-hangup behavior.

11. Host Requests Roll-Call

The host can request, either by DTMF input or through the web interface,that an audio roll call be played for the host. The roll call caninvolve playing the audio clips that each participant recorded whenhe/she joined the conference (including the host's own clip). Theattendee-ID will be read before each clip is played. The reading of theattendee-ID is optional—it can be turned off by setting the“x-roll-call-type” header when the conference is created. The defaultcan be set to “standard”, which means the attendee-IDs will be playedduring the roll call. An alternative is “NoParticipantID”, which willsuppress the playing of the attendee-ID. If at any time during theplayback of a roll call the host wants to stop the roll call, an INFOmessage can be sent with a command to stop the roll call. This commandmay receive an input parameter in the general form of an“x-conference-id”.

1. The host enters DTMF signals that represent a request for a rollcall. The VRU 705 sends an INFO message to the CMS 735, via the proxyserver 715, with a header in the general form of x-Command: PlayRollCalland a header in the form of x-conference-id.

2. The proxy server 715 forwards the INFO message to the CMS 735.

3. The CMS 735 checks the conferencing database 725 to validate therequest, as discussed above. In this case, assume the request is valid(e.g., the sender is the host, the conference exists, etc.).

4. The CMS 735 sends the INFO message to the mixer 730 hosting theconference, echoing the headers from the INFO message it received fromthe VRU 705.

5. The mixer 730 sends an INFO Response message to the CMS 735. Themixer 730 then plays the roll call on the RTP stream directed to thehost.

In an alternative embodiment, the request for the roll call does not gothrough the CMS 735. This request could go from the VRU 705 directly tothe mixer 730. Since the request for the roll call does not change thestate of the conference, the CMS 735 need not necessarily process orotherwise be involved with this request. However, since the host mightmake the request through the web interface, which may go through the CMS735, different embodiments of the instant invention can either includeor exclude the CMS 735 in processing this request. Also, including theCMS 735 in the message sequence may enable the CMS 735 to track how manytimes the roll call feature is used.

6. The CMS 735 sends an INFO Response message to the VRU 705 via theproxy server 715 indicating a status of 200 OK.

12. Host Requests Participant Count

The message sequence for requesting a count of participants can beidentical to the one above for requesting a roll call. One difference,in some embodiments of the instant invention, is that the mixer 730 canplay for the host a sound clip that speaks the number of participantsinstead of playing a roll call. A suitable command to realize thisfunction could be in the general form of “x-command:PlayParticipantCount”.

In other embodiments of the instant invention, command could be executedon the VRU 705 without involving the CMS 735 or the mixer 730. In theseembodiments, an application running on the VRU 705 can access theconferencing database 725 directly, retrieve the data representing theparticipant count, and provide the participant count to the host, e.g.,verbally.

13. Host Requests that Conference be Recorded

The voice stream or various voice streams comprising the conferencecould be recorded by an application running on the VRU 705 coupled tothe host or on the mixer 730 hosting the conference. If an applicationrunning the mixer 730 records the conference, the message flow relatedto the record conference can be based on the flow for the roll callrequest. A suitable command may take general form of “x-command:RecordConference”.

14. The CMS 735 Ends Conference

In some circumstances, the CMS 735 may unilaterally end an ongoingconference, for any one of several reasons. In the context of a pre-paidconferencing plan, an ongoing conference may be terminated if theconference exhausts all of the host's existing, stored pre-paid value.In other cases, all participants but one may have hung up, or the hostmay have hung up and the conference is configured to end when the hosthangs up.

In any event, the message flow when the CMS 735 ends a conference isdiscussed above in connection with Section 4. The CMS 735 can use acommand header in the general form of “x-Command: TerminateConference”to command the mixer 730 to end a particular conference. The INFOmessage may also contain the x-conference-id header.

15. Participant Drops Unexpectedly

The mixer 730 can detect if the RTP stream from a participantdisappears. Sometimes this may be normal. For example, the participantcould be in listen-only mode, or even if the participant was notexplicitly placed in listen-only mode, he may have muted himself. Inthis case, the VRU 705 (or SIP phone, etc.) may stop sending RTP packetsto save bandwidth. Therefore, it may not be necessary to take actionwhen these circumstances occur.

16. Mixer Crashes

If the mixer 730 hosting a given conference “crashes”, an application onthe VRU 705 can detect the loss of the RTP stream from the mixer 730,and thus can detect failure of the mixer 730. Likewise, the CMS 735and/or the proxy server 715 can detect the loss of a mixer 730 bymonitoring the heartbeat messages from the mixer 730. The CMS 735 canhandle mixer failure by e.g., requesting the proxy server 715 to createa new conference on another “live” mixer 730, and updating theconferencing database 725 and the VRU 705 state tables accordingly toreflect that the conference is now hosted on the second mixer 730.

17. Mixer Heartbeat Monitoring

The software running on a given mixer 730 can send a heartbeat messageto the CMS 735 after expiration of a pre-defined interval. The intervalat which the message can be specified as a configurable parameterrelated to the mixer 730. Even though the CMS 735 may not itself loadbalance among the mixers 730 (this can be done instead by the proxyserver 715), it may be useful for the CMS 735 to maintain at least someinformation about the state of the mixers 730 at its disposal.Maintaining this information in, e.g., a database 725 can allow for somedegree of system-wide monitoring from the conferencing database 725.

The heartbeat message can be sent as a SIP INFO message from each mixer730, through the proxy server 715, to the CMS 735. This message cancontain a header in the following general form of x-heartbeat:PercentIdle.

Receiving and processing this heartbeat message will allow the CMS 735to store, e.g., in the conferencing database 725, current informationabout which mixers 730 are operative and what their CPU load is. Thistype of information also enables the proxy server 715 to load balanceamong various mixers 730, and to select a given mixer 730 to host a newconference. Other information can be added to the heartbeat messages asdesired.

In other embodiments, the CMS 735 (or other software) can send an INFOmessage to the mixer 730 instructing it to adjust its heartbeat intervalor frequency. This INFO message can contain a header in the followinggeneral form of x-command: SetHeartbeatPeriod n. where n is the numberof seconds between heartbeat messages.

18. Registration Procedures

This section describes an illustrative strategy suitable for registeringinterface servers 720 and mixers 730 with the SIP proxy servers 715.Each mixer 730 can register with the proxy servers 715 based on entriesin the configuration file corresponding to the mixer(s) 730. Theconfiguration file will preferably contain an entry for each proxyserver 715 with which a mixer 730 is to register. In illustrativeembodiments of the invention, a given mixer 730 will register once witheach proxy server 715 at the mixer's local site.

In these illustrative embodiments, interface servers 720 need onlyregister once with each proxy server 715 at the local site. Inalternative embodiments of the invention, interface servers 720 may ormay not register with remote proxy servers 715.

If all mixers 730 at a given site are near capacity, and have thusunregistered themselves with the local proxy servers 715, the CMS 735may need to create new conferences at other sites, if requests for newconferences so demand. For example, the CMS 735 can be configured to“know” about a proxy server 715 located at a remote site. Were there nomixers 730 of a given class registered with the local proxy server 715,the CMS 735 could send an INFO message to a remote proxy server 715 totry to establish a new conference at that remote site. This processingwould preferably be transparent to the VRU 705 requesting the newconference. If this new conference is successfully created at the remotesite, the VRU 705 would receive the 302 redirect response with asuitable header indicating that a remote mixer 730 will be hosting thenew conference, and that the VRU 705 should contact that remote mixer730 to establish a link therebetween.

19. Outdial

Other aspects of the instant invention can include outdialing from agiven conference to a third party.

1. A conferencing application can collect DTMF input from the hostindicating that the host needs to initiate an outdial. The conferencingapplication then seizes a SIP session that can be used in an IP callbetween the host and the outdialed party. The application assigns aninstance string to this SIP session.

2. The conferencing application queues the outdial request into anoutdial database 706. Information in the outdial request can include atleast:

Number to dial

Unit

Line

setup time

SIP instance

3. The outdial VRU 705 c pulls the request from the outdial database706. Before the outdial VRU 705 c dials the destination number, it mustfirst establish the IP connection with the host VRU 705 a. The outdialapplication seizes an IP line and a SIP session and instructs the SIPdto send the INVITE request to the host VRU 705 a.

4. The SIPd on the outdial VRU 705 c sends INVITE to the inbound (host)VRU 705 a to establish an IP-based communication therebetween. SIPd is ahigh performance, scalable SIP proxy and location server, written in,e.g., the C programming language. The SIP URL to which this request issent is constructed from parameters stored in the outdial database 706.

5. The SIPd on the host VRU 705 a receives the INVITE message, and basedon the parameters therein it determines a destination 780 to which tosend a message. SIPd sends an message transfer layer (MTL) message tothis destination to inform an application that it should open the IPconnection between the host VRU 705 a and the outdial VRU 705 c. Thetransport information that was received in the INVITE message isincluded in the MTL message, along with an SIP instance parameter.

6. The application sends a generic MTL message to the IP daemon to seizean IP channel. As part of this process, the application determines thelocal transport parameters for the RTP stream.

7. The application sends a generic MTL message to SIPd with theparameters (the local transport parameters) that should be used to sendthe 200 OK message to the sip daemon on the outdial VRU.

8. SIPd sends the 200 OK message to the outdial VRU. Outdial VRU SIPdreceives the 200 OK and sends an MTL message to inform the outdial appthat it should open the IP connection.

9. Outdial VRU opens the RTP stream and sends the ACK back to the hostVRU. The application on the host VRU opens the RTP channel and routesthe audio so the caller (host?) is now listening to the audio from theRTP stream.

10. After sending the ACK message to the host VRU, the outdial VRUroutes the audio and dials the outbound number. The IP call between thehost and the dialed party is established.

11. The host enters DTMF indicating he wants to bring the called partyinto the conference. Application makes library call, which sends ageneric message to SIPd telling it to send an INFO message to theoutdial VRU. The INFO message contains special x-headers. Anillustrative format for this information can include the following:

-   -   x-command: JoinConference    -   x-participant-code: 12345 (the participant code that the outdial        VRU must include in its message can take the following form:        INVITE message to the CMS 735)

12. The SIPd on the outdial VRU gets this INFO message and passes it tothe application in an MTL message. The application on the outdial VRUcan adopt any of the following embodiments. In one embodiment, it canprompt the dialed party to ask whether he or she wants to join theconference. If the dialed party decides that it wants to join theconference, that party—or a VRU application interacting with thatparty—can use a variant of the above message flow to enable the calledparty to join the conference. This variant of the above message flow canbe identical to how any caller joins the conference, except instead ofprompting the dialed party for a conference code or participant code,this variant uses the participant code that is passed as a parameter inthe illustrative INFO message given above. This results in an INVITEmessage being sent to the CMS 735, and the outdialed party can joins theconference similarly to any other caller 700, more particularly aparticipant. Preferably, the application on the outdial VRU record aname clip for the outdialed party.

13. The IP session between the two VRU is cleaned up. The host VRU sendsa BYE to the outdial VRU, and both sides release the IP and SIPsessions. The host VRU then routes the audio back to the conference IPsession.

14. The host enters DTMF signals indicating he wants to rejoin theconference but NOT invite the outdialed party in with him. The outdialapplication calls a library function to end this SIP session. TheLibrary can then send a generic message to the SIPd to send the BYEmessage and release the SIP session.

Illustrative SIP Header Extension Fields

This section summarizes the various headers used in various embodimentsof the invention. These headers are referenced above in the discussionof the above illustrative message flows. INFO messages can contain anappropriate command header to indicate the purpose of the INFO message.Table 2 lists various illustrative but non-limiting command values.Following table 2, the headers are defined in table 3. Some headers canbe used in commands (i.e. INFO messages) or used in the INVITE command.

Table 2 contains descriptions of the various commands referenced herein.

TABLE 2 x-command Descriptions x-command value Description Relatedheaders TerminateConference Sent from CMS 735 to x-conference-id Mixer730 to end a x-terminate-reason conference. MuteAll Sent from VRU 705 tox-conference-id CMS 735 and CMS 735 to Mixer 730. Causes all conferenceparticipant talk privileges to be turned off. Can be followed by the Un-mute command for selected participants. Un-muteAll Sent from VRU 705 tox-conference-id CMS 735 and CMS 735 to Mixer 730. Causes all conferenceparticipant talk privileges to be turned on. Mute Sent from VRU 705 tox-conference-id CMS 735 and CMS 735 to x-attendee-id Mixer 730. Causes aspecific participant's talk privileges to be turned off. Un-mute Sentfrom VRU 705 to x-conference-id CMS 735 and CMS 735 to x-attendee-idMixer 730. Causes a specific participant's talk privileges be turned on.SetEntry Sent from VRU 705 to x-conference-id CMS 735 and CMS 735 tox-entry Mixer 730. Specifies the desired state of entry name/ tones. Canbe used to change parameters from how they were set at conferencecreation. SetExit Sent from VRU 705 to x-conference-id CMS 735 and CMS735 to x-exit Mixer 730. Specifies the desired state of exit name/tones. Can be used to change parameters from how they were set atconference creation. LockConference Sent from VRU 705 to x-conference-idCMS 735. No new participants can be added to the conference once it islocked. UnlockConference Sent from VRU 705 to x-conference-id CMS 735.RemoveParticipant Sent from VRU 705 to x-conference-id CMS 735 and CMS735 to x-attendee-id Mixer 730. Causes the mixer 730 to hang up on thespecified participant. SetWrapTime Sent from VRU 705 to x-conference-idCMS 735. This command is x-seconds used to update the value from how itwas set upon conference creation. SetRollCallType Sent from VRU 705 tox-conference-id CMS 735 and CMS 735 to x-roll-call-type Mixer 730.Specifies whether or not an attendee ID is read preceding each name in aroll call. By default, attendee IDs are read. PlayRollCall Sent from VRU705 to x-conference-id CMS 735 and CMS 735 to Mixer 730. Causes mixer730 to play the roll call to the host. StopRollCall Sent from VRU 705 tox-conference-id CMS 735 and CMS 735 to Mixer 730. Stops playing rollcall (interrupts it). PlayParticipantCount Sent from VRU 705 tox-conference-id CMS 735 and CMS 735 to Mixer 730. Causes mixer 730 toannounce the current participant count to the host. StartRecording Sentfrom VRU 705 to x-conference-id CMS 735 and CMS 735 to Mixer 730. Causesthe mixer 730 to immediately begin recording the current conference.StopRecording Sent from VRU 705 to x-conference-id CMS 735 and CMS 735to Mixer 730. Causes the mixer 730 to immediately stop recording thecurrent conference. SetHeartbeatPeriod Sent from an administrativex-seconds tool to Mixer 730 or from CMS 735 to Mixer 730. Changes thenumber of seconds between heartbeat messages. CreateNewConference Sentfrom CMS 735 to x-conference-id Mixer 730 to create a newx-on-hold-music-url conference with the x-periodic-message- specifiedconfiguration. url x-upsell-message-url x-entry x-exitx-lecture-only-flag x-record-flag x-roll-call-type x-max-hold-timex-max-wrap-time WrapWarning Sent from CMS 735 to x-conference-id Mixer730. Causes the x-seconds mixer 730 to play a warning message to theconference warning that the conference will end in the specified numberof seconds because the host has left. HoldWarning Sent from CMS 735 tox-conference-id Mixer 730. Causes the x-seconds mixer 730 to play awarning message to the conference warning that the conference will endin the specified number of seconds unless the host arrives. StatusUpdateSent from VRU 705 to x-conference-id CMS 735 or Mixer 730 tox-attendee-id CMS 735. Informs CMS x-status-change 735 of a change thatoccurred. CMS 735 may already “know” about the change, this message is aconfirmation. Heartbeat Sent from Mixer 730 to x-cpu-idle CMS 735.Informs CMS 735 of the current mixer CPU load.

Table 3 contains a description of various header fields referencedherein, such as in Table 2 above.

TABLE 3 x headers used Header Description/Value x-host-code The hostcode. Sent in INVITE message from VRU when conference host calls in.x-mixer-group Sent from VRU to CMS in the INVITE message. Instructs CMSabout which mixer classes to try to create the conference on, in orderof priority. x-terminate-reason The reason why a conference is beingterminated. Sent from CMS to the mixer in the x-command:TerminateConference message. Value is one of: Hold, Wrap, Normal.x-participant-code The participant code. Sent in INVITE message from VRUwhen conference participant calls in. x-nameclip-url Sent from VRU tothe mixer in the INVITE message. x-client The client name. Sent inINVITE message from VRU. x-mixer-group A pipe-delimited string listingmixer classes that orchestra will try, in the order it would try them.Sent in INVITE message from VRU. x-site The VRU site name (e.g. Denver,Atlanta). Sent in INVITE message from VRU. x-email-notify Y or N. Sentin INVITE message from VRU. x-email-address Email address to use forconference notification. Sent in INVITE message from VRU. x-phone TheANI of the caller (who called into the VRU). Sent in INVITE message fromVRU. x-app-tag Passed from VRU to CMS in the INVITE message to identifythe application. x-dialed-number Passed from VRU to CMS in the INVITEmessage to identify the number dialed by the caller. x-max-duration Usedfor pre-paid conferencing, to specify how long conference is permittedto proceed. x-end-prompt-time Used for pre-paid conferencing to specifya time at which a prompt related to the expiration of a conference willbe issued. x-attendee-id A string of digits or integer that identifies aconference participant uniquely within a given conference. Assigned byCMS and included in the INVITE message sent from the VRU to the mixer.Preferably unique per conference. This number will be read precedingeach name clip for a roll call. The header is also sent in various INFOmessages, because it is used to reference a participant for an action.(mute/ un-mute, delete, etc.) x-conference-id A globally unique stringmade up by CMS to identify a conference. Sent in the INVITE message fromVRU to Mixer, as well as in most command (INFO) messages.x-participant-type Host, Participant, Lhost, Lparticipant (L forlecturer, applies only to lecture only conference). Sent in the INVITEMessage from the VRU. x-seconds Used with various commands to specify atime value. x-status-change Used with the StatusUpdate command. Will beone of: JoinedConference LeftConference Muted Un-mutedx-on-hold-music-url A URL of a file to use for the music track. Use thestring “standard” to specify standard music. Use “none” to create aconference with no on hold music. If the header is not present, thedefault (standard) music will be used. Sent along with theCreateNewConference command. Also sent in INVITE messages from VRU.x-periodic-message-url A URL of a file to use for the periodic on-holdmessage. Use the string “standard” to play a standard message. Use thestring “none” to specify that no periodic message be played. Willdefault to standard periodic message if header not present. Sent alongwith the CreateNewConference command. Also sent in INVITE messages fromVRU. x-roll-call-type Specifies whether or not roll calls will containthe attendee id clips. Default is to play the attendee id clips. To turnthem off, use “NoAttendeeID”. To explicitly turn them on, use“Standard”. Sent along with the CreateNewConference command and also inthe SetRollCallType command. Also sent in INVITE message from VRU.x-upsell-message-url A URL of a file to play once to a participant whileholding. Will default to no upsell message if header is not present.Sent along with the CreateNewConference command. Also sent in INVITEmessages from VRU. x-entry One of: None, Name, Tone, NameAndTone. Willdefault to Tone if header not present. Sent along with theCreateNewConference command. Also sent with the SetEntry command. AndINVITE messages from VRU. x-exit One of: None, Name, Tone, NameAndTone.Will default to Tone if header is not present. Sent along with theCreateNewConference command. Also sent with the SetExit command andINVITE messages from the VRU. x-lecture-only-flag Y or N. Will defaultto N if header is not present. Sent in CreateNewConference command andINVITE messages from VRU. x-record-flag Y or N. Will default to N ifheader is not present. Sent in CreateNewConference command and INVITEmessages from VRU. x-max-hold-time Max # of seconds to allowparticipants to hold for a conference while waiting for the host toarrive. Value 0 means can't hold at all, host must join first. Willdefault to 600 seconds if header is not present. Sent inCreateNewConference command and INVITE messages from VRU.x-max-wrap-time Max # of seconds to allow participants to continue onconference after host leaves. Value of 0 indicates conference terminatesimmediately when host leaves. Will default to 1200 seconds if header isnot present. Sent in CreateNewConference command and INVITE messagesfrom VRU. x-cpu-idle Sent in the Heartbeat command from the Mixer toCMS.Registration & Load Balancing

The invention provides alternative embodiments described below forperforming registrations and load balancing for various SIP devices.

Case 1: No Use of Layer 3 (“L3”) Switching

With Which Proxy WHO WHAT Servers 715 to Register: VRU 705sip:sipd@eosxx.url Local proxy servers 715 only. CMS 735sip:orchestra@siteyy.url Local proxy servers 715 only by default. Remoteproxy servers 715 when needed to create conference on remote mixerbecause local mixers are unavailable for new conference. Mixer 730sip:mixerA@wic.url Local proxy servers 715 only. sip:mixerB@wic.url

1. The VRU 705 can be configured to register with the local proxyservers 715 only. The VRU 705 registers with a proxy server 715 so thatit will receive OPTIONS messages from the proxy servers 715. The VRU 705can respond to these options messages using the X-Load header so thatthe proxy server 715 keeps sending them. The VRU 705 will keep a list ofproxy servers 715 that are active and round robin amongst them for eachtransaction or session, assuring that retransmissions for any message goto the same proxy server 715. When the VRU 705 stops getting OPTIONSmessages from any proxy server 715, it assumes that the proxy server 715is down. The VRU 705 may then attempt to re-register with that proxyserver 715 periodically until the proxy server 715 comes back up.

2. On start-up or boot-up, the CMS 735 can register only with localproxy servers 715. The CMS 735 can provide its site code as part of thename under which it registers. This means whenever the VRU 705 or amixer 730 sends a message to the CMS 735, they must send it to the“local” CMS 735 by using the local sitecode in the message. When the CMS735 needs to create a new conference, it first tries sip:mixerA@url. Ifthat fails (no 200 OK response to the CreateNewConference INFO message)then the CMS 735 tries again using sip:mixerB@url. If that fails too,then the CMS 735 will register with remote proxy servers 715. The CMS735 will then repeat the foregoing process with the remote proxy servers715, thus, the CMS 735 will attempt to register with sip:mixerA@url at aremote site, followed by mixerB@url. The CMS 735 will handle the OPTIONSmessages sent by the proxy server 715 exactly as outlined above for theVRU 705 (failure detection, re-registration).

3. The mixer 730 need only register with local proxy servers 715. Themixer 730 will respond to the OPTIONS messages using the X-Load headerto report its load to the proxy server 715. The following formula can beused to determine the mixer's load:

-   -   Cavg=Average number of participants expected in a conference.        Can be set to e.g., 6.    -   numconf=number of (non empty) conferences existing on the mixer        730 at a given time.    -   numparties=number of participants in a conference at a given        time.    -   M=Maxcalls=absolute maximum number of actual calls that a mixer        730 will accept−can be set to e.g., 240.    -   P=Projected Calls=Cavg*num_conf+SUM over all conferences of        (numparties−Cavg)    -   T=Threshold for new conference acceptance=percentage of maxcalls        that will be allowed. Can be set to, e.g., 75%, which        corresponds to 180 PROJECTED calls.    -   L=Load that will be reported to proxies=P/M

When deciding whether or not to accept a new conference the mixer 730can use the following function: if (L<T) then accept else reject. If themixer 730 is configured to unregister with the proxy server 715 whenL≧1, then the fact that the mixer 730 remains registered with the proxyserver 715 at a given time inherently suggests that the mixer 715 is notoverloaded at that time. This calculation is implicit because the mixer730 unregisters when L>=T)

Case 2: Using L3 Switching

Alternatively, the invention can use L3 switching as a front end for theproxy server(s) 715, and the switches performing the L3 switching willinherently be aware that requests from a given client always use thesame proxy server 715. If this is the case:

WHO WHAT WHERE VRU 705 NO REGISTRATION N/A NEEDED CMS 735sip:orchestra@url Local proxy server 715 only. Mixer 730 sip:mixerA@urlLocal proxy server 715 only. sip:mixerB@url

1. The VRU 705 need not to register with any proxy server 715. In thisembodiment, the VRU 705 is configured to use the L3 switch as the proxyIP address, and the intelligence to perform the functions that wouldotherwise require the VRU 705 to register with the proxy server 715becomes intrinsic to the switch.

2. On boot, the CMS 735 need not register only with local proxy servers715. The CMS 735 need not use its site code as part of the registrationname, because it can access remote proxy servers 715 without registeringwith them. It need only be configured with the IP address of the remoteproxy server 715 (L3 switch). When the CMS 735 needs to create a newconference, it first tries sip:mixerA@url. If that fails (no 200 OKresponse to the CreateNewConference INFO message) then the CMS 735 triesagain using sip:mixerB@url. If that fails too, then the CMS 735 will tryagain with mixerA using the remote proxy address. The CMS 735 willhandle the OPTIONS messages sent by the proxy server 715 by respondingwith the Load header.

3. The mixer 730 will register only with the local proxy servers 715.The mixer 730 will respond to the OPTIONS messages from the proxy server715 using the Load header to report its load to the proxy server 715.The following formula is used to determine the load of a mixer 730:

Cavg=Average number of participants expected in a conference, can be setto e.g., 6.

numconf=number of (non empty) conferences existing on the mixer 730 at agiven time.

numparties=number of participants in a conference at a given time.

M=Maxcalls=absolute maximum number of actual calls that mixer 730 willaccept—can be set to e.g., 240.

P=Projected Calls=Cavg*num_conf+SUM over all conferences of(numparties−Cavg)

T=Threshold for new conference acceptance=percentage of maxcalls thatwill be allowed. Will be set to e.g., 75%, which corresponds to 180PROJECTED calls.

L=Load that will be reported to proxies=P/M

When deciding whether or not to accept a new conference the mixer 730can use the following function: if (L<T) accept else reject.

1.1 Mixer Classes

As shown above, the mixer 730 will register using a SIP url of the formsip:<mixer-class>@url. The invention can include defining differentclasses of mixers 730, with each class having specific performanceparameters or characteristics associated with mixers 730 registeredunder that class. The class under which a given mixer 730 is to registeris specified in the mixer configuration file.

For example only, an illustrative configuration can have 4 mixers 730register as, e.g., a type “A” mixer 730 and 1 mixer 730 register as,e.g., a type “B” mixer 730. Further, the invention can include providingone or more special mixer classes defined for specific clients that canonly be used by that client. The VRU 705 can use a database lookup toaccess a list of mixers 730 in the appropriate class that the CMS 735should use to create the conference. The CMS 735 will first try thefirst element in the list. If it fails it will try the 2^(nd) element,then 3^(rd), etc. The VRU 705 will first use the host code to look up a“Level of Service” parameter applicable to that host. This level ofservice is used to look in another database table to look up the mixer730 class string. The mixer 730 class string will then be used by theVRU 705 in an additional header in the INVITE message sent to the CMS735. The header can be in the form of x-mixer-group, and the format canbe a pipe-delimited list of mixer classes, e.g., x-mixer-group: A|BC|Sams.

One or more suitable application programs can implement softwareresiding on a computer-readable medium or media and embodying thevarious aspects of the method discussed herein and shown in the drawingfigures. This software can be coded using any suitable programming orscripting language. However, it is to be understood that the inventionas described herein is not dependent on any particular operating system,environment, or programming language. Illustrative operating systemsinclude without limitation LINUX, UNIX, or any of the Windows™-family ofoperating systems, and illustrative languages include without limitationa variety of structured and object-oriented languages such as C, C++,Visual Basic, or the like. Further, various aspects of the teachingherein may be implemented in hardware or firmware, instead of or inconjunction with the software embodiments discussed above.

As those skilled in the art will understand, the program of instructionscan be loaded and stored onto a program storage medium or device (notexplicitly shown) readable by a computer or other machine, embodying aprogram of instructions executable by the machine to perform the variousaspects of the invention as discussed and claimed herein, and asillustrated in the drawing figures. Generally speaking, the programstorage medium can be implemented using any technology based uponmaterials having specific magnetic, optical, semiconductor or otherproperties that render them suitable for storing computer-readable data,whether such technology involves either volatile or non-volatile storagemedia. Specific examples of such media can include, but are not limitedto, magnetic hard or floppy disks drives, optical drives or CD-ROMs, andany memory technology based on semiconductors or other materials,whether implemented as read-only, flash, or random access memory. Inshort, this embodiment of the invention may reside either on a mediumdirectly addressable by the computer's processor (main memory, howeverimplemented) or on a medium indirectly accessible to the processor(secondary storage media such as hard disk drives, tape drives, CD-ROMdrives, floppy drives, or the like). Consistent with the above teaching,program storage device 30 can be affixed permanently or removably to abay, socket, connector, or other hardware provided by the cabinet,motherboard, or other component of a given computer system.

Those skilled in the art will also understand that a computer programmedin accordance with the above teaching using known programming languagesprovides suitable means for realizing the various functions, methods,and processes as described and claimed herein and as illustrated in thedrawing figure attached hereto.

Those skilled in the art, when reading this description, will understandthat unless expressly stated to the contrary, the use of the singular orthe plural number herein is illustrative, rather than limiting, of theinstant invention. Accordingly, where a given term is discussed in thesingular number, it will be well understood that the invention alsocontemplates a plural number of the item corresponding to the given termand vice versa, unless expressly stated herein to the contrary.

Those skilled in the art will further recognize that for the purposes ofconvenience, legibility, and clarity, the drawings illustrate variousdata stores 710 and 725 separately, and they are discussed separatelyherein. However, the embodiments shown herein are illustrative ratherthan limiting, and some or all of these various data stores couldreadily be combined or consolidated into one or more data stores, orcould be further subdivided into smaller data stores, without departingfrom the scope of the invention.

The term “data store” herein refers to any storage medium capable ofstoring data, whether realized using semiconductor, magnetic, or opticaltechnology. This term can also include abstract data structuressupported by any number of programming languages, with non-limitingexamples including queues, stacks, linked lists or the like, all ofwhich are implemented at the machine level by disk storage,semiconductor memory, optical media, or the like. If the data store isimplemented as a database, this database can take the form of arelational database, an object-oriented database, and any combinationthereof, or any other known database technology. Suitable databaseserver programs are readily available from a variety of vendors,including IBM/Informix, Microsoft, Oracle, or the like.

Various embodiments of the invention are described above to facilitate athorough understanding of various aspects of the invention. However,these embodiments are to be understood as illustrative rather thanlimiting in nature, and those skilled in the art will recognize thatvarious modifications or extensions of these embodiments will fallwithin the scope of the invention, which is defined by the appendedclaims.

1. A conferencing system comprising at least the following: at least one voice response unit adapted to interact via a circuit-switched network with a plurality of callers, the voice response unit further adapted to support at least one conferencing application and at least one non-conferencing application; at least one mixer in communication with the voice response unit via a packet-switched network, the at least one mixer adapted to support at least one conference call between at least two callers communicating with one another via the mixer and the VRU; at least one data store adapted to store data representing at least one state parameter relating to at least one conference call supported by the mixer, the data store coupled to communicate with the VRU and the mixer.
 2. The system of claim 1, further comprising at least a second VRU in communication with the mixer, and wherein the mixer is adapted to support a conference call including a first caller communicating via at least the first VRU and at least a second caller communicating via the second VRU.
 3. The system of claim 2, further comprising at least a third VRU in communication with the mixer, and wherein the mixer is adapted to support a conference call including at least a third caller communicating via the third VRU.
 4. The system of claim 1, wherein the VRU is adapted to execute the conferencing application in response to recognizing at least a first destination telephone number dialed by at least a first caller, and wherein the VRU is adapted to execute the non-conferencing application in response to recognizing at least a second destination telephone number dialed by at least a second caller.
 5. The system of claim 1, wherein the VRU is adapted to execute the conferencing application in response to recognizing at least a first DNIS parameter associated with at least a first caller, and wherein the VRU is adapted to execute the non-conferencing application in response to recognizing at least a second DNIS parameter associated with at least a second caller.
 6. The system of claim 1, further comprising at least one interface server coupled between the data store and the VRU, and adapted to queue requests from at least the VRU for processing by the data store.
 7. The system of claim 1, further comprising at least a second interface server coupled between the data store and the VRU, and adapted to queue requests from at least the VRU for processing by the data store.
 8. The system of claim 1, further comprising a proxy server coupled to the VRU to receive conferencing-related requests from at least the VRU.
 9. The system of claim 8, wherein the proxy server is adapted to communicate a request to create a new conference to the mixer.
 10. The system of claim 1, further comprising at least one proxy server coupled to the VRU to receive conferencing-related requests from at least the VRU.
 11. The system of claim 1, further comprising at least one interface server coupled between the data store and the VRU and adapted to queue requests from at least the VRU for processing by the data store.
 12. The system of claim 1, further comprising at least a second mixer in communication with the VRU.
 13. The system of claim 12, further comprising a proxy server coupled to the VRU to receive conferencing-related requests from at least the VRU, and wherein the proxy server is adapted to submit a request to create a new conference to at least one of the mixer and the second mixer.
 14. The system of claim 1, further comprising a provisioning database in communication with the VRU, the provisioning database being responsive to a conference code received from the VRU to provide a signal to the VRU indicating whether the conference code is valid.
 15. The system of claim 1, further comprising a plurality of interface servers coupled between the data store and the VRU and adapted to queue requests from at least the VRU for processing by the data store, and further comprising a proxy server coupled between the VRU and the plurality of interface servers, and wherein the proxy server is adapted to perform load balancing among the interface servers.
 16. The system of claim 1, wherein the mixer has associated therewith a capacity parameter that is determined on a basis other than a number of discrete hardware ports associated with a given mixer.
 17. The system of claim 1, wherein the mixer does not include discrete hardware ports.
 18. The system of claim 1, wherein the mixer has a variable capacity.
 19. The system of claim 1, wherein the mixer has a non-fixed capacity.
 20. The system of claim 1, wherein the mixer comprises general-purpose server hardware executing at least one application program adapted to at least to mix voice streams originating from respective conferees associated with at least one given conference.
 21. The system of claim 1, wherein at least the VRU, the mixer, and the data store are coupled to communicate at least indirectly via a local area network.
 22. The system of claim 1, wherein at least the VRU, the mixer, and the data store are coupled to communicate at least indirectly via a network employing a voice-over-IP protocol.
 23. The system of claim 1, wherein at least the VRU, the mixer, and the data store are coupled to communicate at least indirectly via a packet-switched network. 